The present invention is related to application entitled METHOD AND APPARATUS FOR AUTOMATIC TRANSFER OF A CALL IN A COMMUNICATIONS SYSTEM IN RESPONSE TO CHANGES IN QUALITY OF SERVICE, Ser. No. 09/358,994, filed even date hereof, and assigned to the same assignee.
1. Field of the Invention
The present invention relates generally to communications system and in particular to a method and apparatus for routing calls in a communications system. Still more particularly, the present invention relates to a method and apparatus for routing voice over Internet protocol calls within a communications system.
2. Background of the Invention
Originally regarded as a novelty, Internet telephony is attracting more and more users because it offers tremendous cost savings relative to the traditional public switch network (PSTN) users can bypass long distance carriers and their permanent usage rates and run voice traffic over the Internet for a flat monthly Internet access fee. Internet telephony involves the use of voice over Internet protocol also referred to as xe2x80x9cvoice over IPxe2x80x9d or xe2x80x9cVoIPxe2x80x9d. This protocol is packet based in contrast to the switch circuit system in a PSTN.
For example, user A in Austin wants to make a point-to-point phone call to user B in the company""s London office. User A picks up the phone and dials an extension to connect with the gateway server, which is equipped with a telephony board and compression-conversion software ;the server configures the PBX to digitize the upcoming call. User A then dials the number of the London office, and the gateway server transmits the (digitized, IP-packetized) call over the IP-based wide area network (WAN) to the gateway at the London end. The London gateway converts the digital signal back to analog format and delivers it to the called party. With this calling system, expensive international long distance charges are virtually eliminated because the call is set up as a local call.
Users of communications system are increasingly mobile and require reliability in calls, such as, business calls. For example, user A may want to continue the conversation with the called party in London, but has to leave for an appointment. In such a situation, user A must terminate or hang up the voice over IP call and reinitiate the call on user A""s mobile phone by redialing the called party""s number. In another example, user A calls a party on a mobile phone while in transit to work. When user A reaches work, user A must hang up the call and redial the called party""s number to start a new call, using voice over IP. In this manner, user A reduces costs for the call.
In addition, the level of reliability and sound quality expected by users is not always available with voice over IP calls. This situation is primarily caused by bandwidth limitations that lead to packet loss in the network. When congestion occurs, delays in packet transmission may occur, resulting in packets being lost or discarded. This packet loss causes gaps or periods of silence in the conversation between users. These gaps or periods of silence lead up to a xe2x80x9cclipped-speechxe2x80x9d effect. Such a situation is unsatisfactory for most users and is unacceptable in business communications. As a result, when a user is dissatisfied with the quality of a voice over IP call, the user must hang up the call and redial the called party""s number to initiate a new call using a legacy phone to continue the conversation with the called party.
Terminating and reinitiating calls in this manner is inconvenient for a caller. As a result, a caller may often times continue a call using a legacy phone, such as a mobile phone, rather than hanging up the legacy phone and redialing the called party""s number on a terminal using voice over IP. Therefore, it would be advantageous to have an improved method and apparatus for allowing a user to take advantage of voice over IP without the user having to terminate a call in progress and redial a called party""s number to initiate a new call to continue the conversation.
The inconveniences to a user desiring flexibility and mobility in a communications system providing calls over a packet based network, such as voice over IP, are minimized through the method and apparatus of the present invention. A request is received from a user, at a first terminal in a communications system, during a call to switch the call from a packet based network to a circuit switched network. The call is switched to a second terminal associated with the user in response to receiving the request. The second terminal uses the circuit switched network and the call is switched to the second terminal without terminating the call.
The present invention also provides for switching from a path in a circuit switched network to a packet based network in response to a request from a user. When a request is received from a terminal during the call, the call is switched to another terminal using the packet based network.
The switching of the call between a packet based network and a circuit switched network may be accomplished by establishing a new path to a new terminal on the desired network while the path through the present network continues to be used for the call. When the new path is established, the new path is joined to the call. The portion of the current path through the current network is released or discontinued. The joining of the paths may be accomplished through a call conferencing feature used to provide call conference functions. The destination for the call may be selected by associating the user with a preselected destination stored in a database, which is queried when the user makes a request to switch or transfer the call.
Other aspects and features of the present invention will become apparent to those ordinarily skilled in the art upon review of the following description of specific embodiments of the invention in conjunction with the accompanying figures.